In mobile communications, it is necessary to compress and encode digital information such as speech and images for efficient use of radio channel capacity and storage media for radio waves, and many coding and decoding schemes have been developed so far.
Among these, the performance of speech coding technology has been improved significantly by the fundamental scheme of “CELP (Code Excited Linear Prediction),” which skillfully adopts vector quantization by modeling the vocal tract system of speech. Further, the performance of sound coding technology such as audio coding has been improved significantly by transform coding techniques (such as MPEG-standard ACC and MP3).
In speech signal coding based on the CELP scheme and others, a speech signal is often represented by an excitation and synthesis filter. If a vector having a similar shape to an excitation signal, which is a time domain vector sequence, can be decoded, it is possible to produce a waveform similar to input speech through a synthesis filter, and achieve good perceptual quality. This is the qualitative characteristic that has lead to the success of the algebraic codebook used in CELP.
On the other hand, a scalable codec, the standardization of which is in progress by ITU-T (International Telecommunication Union—Telecommunication Standardization Sector) and others, is designed to cover from the conventional speech band (300 Hz to 3.4 kHz) to wideband (up to 7 kHz), with its bit rate set as high as up to approximately 32 kbps. That is, a wideband codec has to even apply a certain degree of coding to audio and therefore cannot be supported by only conventional, low-bit-rate speech coding methods based on the human voice model, such as CELP. Now, ITU-T standard G.729.1, declared earlier as a recommendation, uses an audio codec coding scheme of transform coding, to encode speech of wideband and above.
Patent Document 1 discloses a scheme of encoding a frequency spectrum utilizing spectral parameters and pitch parameters, whereby an orthogonal transform and coding of a signal acquired by inverse-filtering a speech signal are performed based on spectral parameters, and furthermore discloses, as an example of coding, a coding method based on codebooks of algebraic structures.    Patent Document 1: Japanese Patent Application Laid-Open No. HEI10-260698